Wireless audio transmission system and method

ABSTRACT

A system and method transmit a data stream from a source to a destination over a communication channel. A transmitter includes devices for processing inputs to assemble data packets for the data stream, and a multiplexer for assembling a data frame to be transmitted over the communication channel, in which each data frame has at least one fixed slot. The multiplexer sets at least one freely allocatable time slot in each data frame. Retransmission control devices connected to the multiplexer retransmit a specific data packet which is not properly received by the destination, using one of the freely allocatable slots.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a 371 of international application numberPCT/NL2004/000225, filed on April 5, 2004.

FIELD OF THE INVENTION

The present invention relates to a method for transmitting at least onedata stream from a source to at least one destination over acommunication channel, the at least one data stream comprising asequence of a plurality of data packets, the data stream beingtransmitted over the communication channel using a protocol comprisingdata frames, in which each data frame has at least one fixed slot.

In a further aspect, the present invention relates to a transmitterarrangement for use in a system for transmitting at least one datastream from a source to at least one destination over a communicationchannel, the transmitter arrangement comprising input processing meansfor assembling data packets for each of the at least one data stream,and a multiplexer arranged to receive data packets for each of the atleast one data stream and to assemble a data frame to be transmittedover the communication channel, in which each data frame has at leastone fixed slot.

In an even further aspect, the present invention relates to a receiverarrangement for receiving at least one data stream transmitted by atransmitter arrangement according to the present invention.

PRIOR ART

A communication system allowing transmission of audio data streams isknown from American patent U.S. Pat. No. 6,373,842, describingunidirectional streaming services in wireless systems. In this method,it is possible for a receiver of data packets to request aretransmission of a data packet which was incorrectly received. Thismethod is suitable for the transmission of audio over a given wirelessaccess network. It does however not provide a solution for sharing ofthe medium by multiple users, or sharing a stream by multiple receivers,where each has specific radio propagation conditions.

The IEEE standards 802.11 as well as 802.15.4 incorporate mechanisms forguaranteeing minimum throughput on specific channels, whilesimultaneously reserving the channel for shared (‘free’) access. Withoutconsidering bandwidth, this would make them suitable for transmittingaudio. However, the principal shortcomings of the abovementionedprotocols, for the purpose of transmitting audio, are, firstly, thatthey are not capable of retransmission control for one data stream withmultiple destinations, secondly, that the fixed (reserved or guaranteed)slots have to be allocated before the start of the frame, whereas ad hocallocation would reduce the probability of exceeding a defined maximumallowed audio latency, and thirdly, that the shared access is notsuitable for audio since it is based on contention with CSMA/CD implyinga loss of throughput and interference robustness.

SUMMARY OF THE INVENTION

The present invention seeks to provide a streaming data communicationsystem, e.g. suited for streaming audio applications, which can operatein an environment with multiple possible sources of interference, whilemaintaining a low latency and a high audio quality.

According to the present invention, a method according to the preambledefined above is provided, in which each data frame further comprises atleast one freely allocatable slot, in which a specific data packet,which is not properly received by the at least one destination, isretransmitted from the source using one of the at least one freelyallocatable slot. This method allows to retransmit a packet which is notproperly received due to e.g. interference, thus resulting in a higherchance of a correct and complete data stream arriving at thedestination.

In a further embodiment, the specific data packet for a specific datastream is retransmitted in the fixed slot of that specific data streamin a next data frame. Although retransmission in a next data frame willincrease the data latency, proper dimensioning of the data frames maykeep this in acceptable limits. Allowing a missed data packet to beresent in a next data frame will even make the transmission method morerobust, increasing the chance of a complete error free data streamreception.

The data frame may in a further embodiment comprises a down-link partand an up-link part, each having at least one fixed slot and at leastone freely allocatable slot. This allows for two-way communication, e.g.including message data (e.g. control data, handshake data, etc.) betweenthe source and the destination.

In an even further embodiment, a data packet has a predefined duration,e.g. 250 μsec for the data packet and the acknowledge space, thepredefined duration being small compared to non-transmitting gaps ofpossible interfering sources. In this embodiment, it is ensured thatdata packets fit in between packets of other, possibly interferingsources, such as Bluetooth, Wifi, DECT, etc. The non-transmitting gapsmay in a further embodiment be detected by carrier sense/detecttechniques, and the method further comprises the synchronization of thedata stream transmission to the detected non-transmitting gaps.

The method, in a further embodiment, comprises receiving acknowledgement(positive acknowledgment (ACK) or negative acknowledgement (NACK)) of areceived data packet from the at least one destination, andretransmitting a not properly received data packet within the same dataframe. This will ensure a low latency, when there is only a short delaybetween reception of a packet and transmission of the acknowledgment. Inan exemplary embodiment, each data frame comprises N fixed slots with asingle negative acknowledge sub-slot following each of the N fixedslots, and each of the freely allocatable slots comprises N negativeacknowledge sub-slots following the freely allocatable slot forindicating in which of the N fixed slots a packet was not received. Thisis a very effective acknowledgment mechanism, requiring little overhead.

In an even further embodiment, the acknowledgment comprises apseudo-noise code. This opens up the possibility for using multi-codeNACK in a multi-frequency system. It provides a good adjacent channelisolation, also in the case when one destination is close to the sourceand another destination is further away (near-far problem).

At least one freely allocatable slot may, in a further embodiment, beallocated to transmit a control data message. Control data messages maybe sent randomly when necessary, not in a continuous fashion. A freelyallocatable slot may allow this, in the case of one or moredestinations.

In the case of a multiple access transmission system, a back-offmechanism may be used for transmitting the control data message formultiple access. When a user tries to transmit, but finds that thechannel is not free, it waits for a predetermined time period beforeretrying. This allows to share the transmission capacity with otherusers.

In a further embodiment antenna and/or frequency diversity is applied tomake the transmission method more robust against interference of allkinds. For example, an adaptive frequency selection mechanism may beused to circumvent channels which experience interference. Also, thisembodiment provides a better rejection of multi path distortion and/orinterference.

To obtain a higher throughput rate in an embodiment of the presentmethod, a data packet is compressed before transmission. As analternative, a data packet is transmitted uncompressed in the firsttransmission, and compressed in at least one of the retransmissions.

In a further embodiment, specifically directed at streaming audioapplications, the method further comprises converting received datapackets in an audio signal, replacing a missing data packet by anearlier received data packet, and smoothing the transition between theearlier received data packet and the replaced data packet. This allowsto prevent audible ticks in case of receive errors, increasing themethod quality of reception. As an example, the smoothing may be appliedusing a raised cosine filter function.

To obtain a flat spectrum transmission signal using the present method(e.g. to comply with regulatory standards), the method further comprisesscrambling a retransmitted data packet before retransmission using apseudo-randomly varying scrambling technique, and descrambling theretransmitted data packet upon reception. Furthermore, the method mayfurther comprise integrating multiple retransmitted data packets. As theerror distribution varies in time, this allows gain from integration,and a more robust method as a result. Also, the method according to thepresent method may be protected against eavesdropping by usingencryption of the data packets in the data stream.

To mitigate problems related to near-far problems, the method maycomprise that the source increases its transmission power upondetermination that a data packet is not properly received by thedestination. This may include a closed loop power control. Furthermore,in a further embodiment, the source compares the received signalstrength to a threshold value, and decreases its transmission power by apredefined step if the threshold value is exceeded, and increases itstransmission power by a predefined step otherwise. The threshold valuemay be adaptively controlled by an outer control loop.

The method, as discussed in the embodiments above, relates to datapackets. A data packet comprises a preamble, and/or a header, and/or atleast one packet of control message data, and/or at least one packet ofapplication data from at least one input. Multiple input data packetsand control message data may be multiplexed into a single stream of datapackets.

In a further aspect, the present invention relates to a transmitterarrangement as defined in the preamble above, in which the multiplexeris further arranged to provide at least one freely allocatable slot ineach data frame, and in which the transmitter arrangement furthercomprises retransmission control means connected to the multiplexer andarranged to retransmit a specific data packet, which is not properlyreceived by the at least one destination, using one of the at least onefreely allocatable slot. In further embodiments, the multiplexer and/orretransmission control means are further arranged to execute the presentmethod. The transmitter arrangement may further comprise a compressor, ascrambler for scrambling a retransmitted data packet beforeretransmission using a pseudo-randomly varying scrambling technique, anencryption module for encrypting data packets before transmission, or amodulator.

In an even further aspect, the present invention relates to a receiverarrangement for receiving at least one data stream transmitted by atransmitter arrangement according to the present invention. The receiverarrangement may further comprise audio processing electronics, adescrambler for descrambling a retransmitted data packet upon reception,and/or a pre-detection accumulator for integrating multipleretransmitted data packets.

SHORT DESCRIPTION OF DRAWINGS

The present invention will be discussed in more detail below, using anumber of exemplary embodiments, with reference to the attacheddrawings, in which

FIG. 1 shows a simplified diagram of a wireless transmission system inwhich the present invention may be applied;

FIG. 2 shows a schematic diagram of the structure of a data frame asused in an embodiment of the present invention;

FIG. 3 shows a flow chart illustrating an embodiment of the methodaccording to the present invention;

FIG. 4 shows an exemplary timeline of an implementation of anacknowledge mechanism in an embodiment of the present invention;

FIG. 5 shows a block diagram of a transmitter according to an embodimentof the present invention;

FIG. 6 shows a block diagram of a receiver according to an embodiment ofthe present invention; and

FIG. 7 shows a simplified block diagram of an embodiment of an errorconcealment module of a receiver according to the present invention.

DETAILED DESCRIPTION OF EXEMPLARY EMBODIMENTS

The present invention relates to a wireless digital audio transmissionsystem, and can be applied in a number of streaming data applications,and is specifically suited for audio data streams. Examples of suchapplications include, but are not limited to, a wireless conferencesystem, a wireless tour guide system, a wireless headphone, wirelessspeakers, wireless stereo audio with MIDI, wireless earphones, awireless microphone, a wireless public address system, a wirelessintercom, etc.

FIG. 1 shows a simplified diagram of an embodiment of the presentinvention, having a single central unit 10, and a number of mobile units12. The central unit 10 is arranged to distribute or collect audio datato/from the mobile units 12. The data streams between the central unit10 and each of the mobile units 12 may be uni-directional (indicated bysingle arrow) or bidirectional (indicated by double arrow, e.g.including bidirectional audio or message data such as requests). Thecentral unit 10 may transmit a data stream addressed to a single mobileunit 12 (uni-cast), several mobile units 12 (group- or multi-cast) orall mobile units 12 (broadcast).

The data streams between the central unit 10 and mobile units 12 aresent by RF signals (air interface), e.g. in the 2400-2483 MHz ISM bandor the 5 GHz UNII/ISM bands. In this embodiment, several data (or audio)streams are transmitted by the central unit 10, using Time DivisionDuplex (TDD) techniques. Data is sent over the air interface in frames,in which each frame comprises time slots for downlink and time slots foruplink. As will be understood, different embodiments may be envisaged,using e.g. only downstream or upstream channels.

A single frame, as used in a particular embodiment, comprising threedownlink audio data streams, and three uplink audio data streams, isdepicted in FIG. 2. The downlink data is organized as broadcast ormulti-cast transmissions, and the uplink data is organized as uni-casttransmissions (always directed to central unit 10). The single frame ise.g. 2 ms long, and comprises a downlink part of 1 ms and an uplink partof 1 ms. If the audio sources are 16 bit PCM coded audio at 24 ksps, andif the air interface speed is 11 Mbps, this results in a latency of lessthan 15 ms, while exhibiting excellent interference robustness. For eachdata stream (or channel), the frame comprises a single fixed time slot(DFX_A . . . DFX_C) in the downlink part and a single fixed time slot(UFX_A . . . UFX_C) in the uplink part. Furthermore, the downlink partcomprises four free time slots (DFR_0 . . . DFR_3) and the uplink partthree free time slots (UFR_0 . . . UFR_2).

In the lower part of FIG. 2, the actual data packets are shown. In thedownlink part, in the fixed time slots, a stream down with data (SDD)packet is followed by a single negative acknowledge (NACK) space. TheNACK packet is sent by the receiving mobile unit 12 in that spacewhenever an expected data stream packet is not received. Using negativeacknowledge packets has certain advantages which will be explainedbelow. In the free slots DFR_0 . . . DFR_3, the downlink data streampacket (SD), which may or may not comprise a resent data packet, isfollowed by three NACK spaces, as any of the three data streams mayrequire a NACK packet from any one of the receiving mobile units 12.

In a generally analogue manner, the uplink channels comprise a stream uprequest (SUR) from one of the mobile units 12 to the central unit 10,followed by a stream up data packet (SUD). In the free uplink timeslots, a stream up request (SUR) is possibly followed by a stream up(SU) packet.

As the frame comprises a combination of fixed allocated time slots andflexibly allocatable time slots, a minimum throughput on a specificaudio data stream is guaranteed, while also bandwidth is available whichmay be used for the audio data streams that need retransmission. Thishas particular advantages for a continuous data source, such as audio,where every air-frame a new data packet is generated. Since the freeslots are shared between multiple streams, a stream with a bad RF linkwill claim most of the free slots, thereby reducing the retransmissioncapacity for other streams. The fixed slot therefore guarantees a streamwith a good RF link to remain unaffected.

A data stream which experiences loss of a data packet (e.g. in time slotDFX_A) will, according to the present invention, retransmit that packetin one of the freely allocatable time slots DFR_0 . . . DFR_3. Thus foreach stream, only one time slot per frame can be fixed, although theremay be more than one retransmission of that data packet in the sameframe. E.g., when a retransmission in time DFR_0 fails, there may be afurther attempt in any of the remaining free time slots DFR_1 . . .DFR_3. If, for a particular stream, the retransmission failsrepetitively, so that a data packet is not successfully transmitted inthe free slots, the fixed slot belonging to this stream and thesubsequent free slots of the next frame can be used to retry sending thedata packet. This process is be repeated a predefined number of frames,until the data packet is considered lost.

The fact that data streams are time division duplexed with fixed andfree slots enables low power consumption, in particular when thereception quality is good. The reason is that a high reception qualityresults in the first transmission attempt of a particular data packet tobe successful. In this case, the fixed slot suffices for transmission ofthe complete audio data stream. Therefore, the receiving device may bepowered-down in between fixed slots of the considered data stream.

In a further embodiment, shared access data transmission is possible inthe present communication system. One or more uplink time slots UFR_0 .. . UFR_2 may be dedicated for shared access, and in these time slots,the central unit 10 actively indicates whether the shared channel isfree or occupied, e.g. using a flag in a request packet SUR. This allowsfor mobile units 12, that are not dedicated to provide an uplink stream,to still send message data to the CU 10. The mobile unit 12 responds byaccessing that particular channel with new message data, but only whenthe channel is indicated as being free. When a first packet of themessage is successfully received by the central unit 10, the flag is setto occupied until the end of the data message reception by the centralunit 10. When a mobile unit 12 tries to access the shared channel, butis unsuccessful, a back-off time is introduced before a further attemptis made by the mobile unit 12 to access the data channel again. By usinga random back-off time, contention by multiple mobile units 12 to accessthe shared channel may be mitigated.

The frame length, and more specifically the time slot/packet length, isvery short to enable successful transmission of a data packet in thepresence of intermittent interfering sources, such as Bluetooth packets,DECT-on-2.4 GHz, wireless LAN transmissions or other non-continuoustransmission sources. In a particular embodiment, the packet length hasa maximum of 250 μs. Such a short packet length imposes strictrequirements, such as a short preamble, short header (an efficientprotocol for medium access control), and a high air interface bit rate.

In the embodiment shown, the central unit 10 dictates the transmissionprotocol in order to prevent packet collisions from the mobile units 12.The central unit 10 may also include a detection method for detectinginterference sources and adjust the timing of its own transmissions tofit between these interference sources. Such a detection method may beimplemented using a carrier sense or carrier detect system.

In order to provide a robust communication system with a low latency anda high audio quality, the present system will resend a data packet whichis not properly received in a free time slot (DFR_0 . . . DFR_3; UFR_0 .. . UFR_2) in the same data frame. This ensures a very low latency inthe data packet stream.

The detection whether a data packet is properly received may be done inone of several manners, known as such in the art, such as parity errordetection schemes, cyclic redundancy checksum (CRC) test, etc. As thesystem may be used for bi-directional communications, in the following,the terms transmitter and receiver will be used, which may be either oneof the central unit 10 or one of the mobile units 12, depending on thetransmission direction.

To maximize the possibility that a missed audio and/or message datapacket can be resent in the same frame, the receiving unit is arrangedto send a (negative) acknowledgment for receiving a packet almostimmediately after reception. In a specific embodiment, an acknowledgment(ACK) must e.g. be received by the transmitting unit within about 80 to200 μs, and a negative acknowledgment (NACK) must be sent by thereceiver within 25 to 60 μs. This feature allows the system to benefitfrom gaps in interference sources. Due to the short packet duration andthe almost immediate ACK/NACK response, there is a high correlationbetween the events that a stream packet is not interfered with and thatthe ACK/NACK response is not interfered, which increases theretransmission efficiency of the present protocol.

The present invention may use any of the automatic retransmissionprotocols (ARQ) which either use a positive acknowledgment (ACK) or anegative acknowledgement (NACK). In a particular embodiment, thereceiver only determines whether RF energy with a specific pseudo-noisesignature is present in the associated NACK slot (see FIG. 2). This isparticularly advantageous in multi-cast or broadcast transmissions,where multiple mobile units 12 may send a NACK response when notproperly receiving a data packet. Although the (multiple) NACK responsesmay interfere with each other, this is not a problem when only detectingthe presence of RF energy.

By using only a single NACK response window in the frame for each of theaudio data streams A . . . C, the protocol is very efficient. Forreliable NACK detection, the fixed slot NACK response window requires aduration of about 20 μs.

In the case of a retransmission in one of the free time slots DFR_0 . .. DFR_3, it must be made clear to the (re)transmitting unit 10, 12 whichaudio data stream A . . . C has been incorrectly received. This may beimplemented in the manner depicted in FIG. 2, where each of the freetime slots DFR_0 . . . .DFR_3 is followed by three NACK windows, eachassociated with one of the fixed audio data streams A . . . . C. When areceiver incorrectly receives an audio data packet for stream A, it willsend a NACK response in the window associated with stream A. Thetransmitter can thus check whether a received NACK response is actuallycorrect by checking the timing.

The present invention may also be applied in multiple frequencytransmission systems, where each frequency carries a protocol asdescribed in the above embodiments. In this case, the frame timing ofthe different frequency signals needs to be synchronized, in order toprevent co-located central units 10 and mobile units 12 at adjacentfrequencies from interfering each other. Still, mobile units 12 that arenear to the central unit 10, may cause interference to mobile units 12in an adjacent frequency channel with a weak link. This is frequentlyreferred to as the near-far effect. This effect will pose a problem toreception of uplink data streams, as well as NACKs. As a solution, themobile unit 12 may apply transmit power control.

Apart from power control, the near-far problem for the NACK responsesmay be further mitigated by coding NACK responses at differentfrequencies with mutually (semi-) orthogonal codes, each frequencyhaving a unique code. Examples of these kind of codes are:

pn05=[1 1 1 1 0]

pn07=[1 1 1 0 1 0 0]

pn11=[1 0 1 1 0 1 1 1 0 0 0]

pn15=[0 0 0 0 1 0 1 0 0 1 1 0 1 1 1]

pn19=[0 0 0 0 1 0 1 0 1 1 1 1 0 0 1 0 0 1 1]

In the case that the interference from adjacent channels experienced isdue to receive chain non-linearity, the high degree of orthogonality maybe lost. The different repetition times of the codes given above maythen be exploited to achieve sufficient adjacent channel isolation.

The detection of NACK responses in the transmitter is advantageouslydesigned for a short NACK length, while still maintaining a highdetection probability, high interference robustness and low false alarmrate. A missed NACK response will result in no retransmission of theassociated data packet, and thus a missing data packet in the audio datastream. When the NACK detector operates based on received RF energy, thedetection period needs to be sufficiently long to be able to collect asignificant amount of energy. According to a specific embodiment of theNACK detector, NACK responses of only 20 μs suffice for detection.

In FIG. 3, a flow chart is shown, in which the flow according to thisspecific embodiment of the present invention is given. In block 2, theNACK receiver is put in its highest gain level, so that even weaksignals are received well above the noise level. While this means thatthe signal may become clipped, it does save the time required for theautomatic gain control loop to settle.

The signal energy level is detected, and in decision block 3, it isdecided whether or not the signal has sufficient power to possibly be aNACK response. If not, the flow returns back to block 2 for the nextNACK time slot detection. If yes, the correlation between samples ismeasured with a time distance equivalent with the symbol rate. Thisallows the receiver to discern between energy from a narrow-bandinterferer and the wide-band NACK response (block 4). If there issufficient narrowband signal energy present (decision block 5), thissignal may be filtered out by using a notch filter. The notch is put ata frequency of the narrowband interferer, which, in one particularembodiment, may be calculated form the phase of the complex correlationsignal (block 6). If no narrowband interference is detected, the flowcontinues directly with block 7, which applies the antenna diversityselection by measuring the energy level at each of the availableantennae, and selects the antenna with the highest energy level. Inblock 8, it is checked whether a NACK response is actually present, e.g.by correlating the received signal (a pseudo noise sequence signal) witha delayed version (delay equal to the PN code repetition time), andcomparing the correlation power with a preset threshold. If a NACKresponse is detected, the transmitter is signaled to resend theassociated data packet (block 9), otherwise the flow returns to block 2.

In FIG. 4, a NACK detector time line of in total 20 μs is illustratedfor the above example. In a first time period, e.g. 3 μs, the totalreceived power is detected, followed by detection and training ofnarrowband interference (also 3 μs). The antenna diversity selection maytake another 4 μs, which leaves a time window of 10 μs for the actualNACK response detection. It is noted that this type of NACK responsedetection, as illustrated in FIGS. 3 and 4, may also be applied in otherARQ protocol implementations.

In all described embodiments, antenna diversity may be used to make thecommunication system more robust against fading. Antenna diversitytechniques may be employed using successive quality measurements onevery preamble of a data packet received, by switching to the antennahaving the best quality index. Alternatively, only a single measurementmay be made on an a priori selected antenna, and switching to the otherantenna may be executed only when the measured quality parameter fallsbelow a predefined threshold.

In FIG. 5, a schematic block diagram is shown of a wireless audiotransmitter arrangement 20 according to an embodiment of the presentinvention. The transmitter 20 may receive message data (message input26, e.g. I2C interface) and one or more audio data streams (audio datainput 21). The audio data input and output may be any type of digitalformat, such as linear PCM, ADPCM, MPEG layer 3 or AC3 compressed audio,which are known in the art. The audio data may be carried over standarddigital audio interfaces, such as S/PDIF, 12S, or IEC61937, which areknown in the art. Each audio source data stream is audio processed(block 22), eg. for sample rate adaptation, and partitioned in audiodata packets (block 23), after which a compression of the audio datapacket may be applied (block 24). Both the audio data packets andmessage data packets are put in a buffer before the data frame (see FIG.2) is assembled. Part of this process is the retransmission control ofany of the (audio) data packets, as described above (retransmissioncontrol blocks 25, 28). The audio and message packets are multiplexedinto a single stream packet in multiplex block 29. Then, the data packetmay optionally be submitted to encryption (block 30) and/or scrambling(block 31), and finally modulation on a carrier frequency (block 32)before being fed to an antenna 33.

FIG. 6 shows a schematic block diagram of an exemplary embodiment of awireless audio receiver 40 according to the present invention. In thereceiver arrangement 40, the various processing blocks of thetransmitter are repeated in an reverse manner: after reception of theair interface signal using antenna 41, the signal is demodulated (block42), descrambled (when necessary, block 43), and decrypted (whennecessary, block 44). The resulting data packet is then de-multiplexedin different audio data packets and message data packets (demux block45). The retransmission control and buffer blocks 46, 51 (audio andmessage, respectively) check whether each data packet is receivedwithout error, and signals when a retransmission is necessary (which isthen taken care of by the transmitter arrangement 20 in the same unit10, 12). For the audio data streams, the packets are then decompressed(block 47), further processed (block 48, de-partitioning and errorconcealment). After that, the audio data streams are audio processed inblock 49, eg. for volume regulation and sample rate adaptation, and fedto the audio I/O (block 50). Message data packets are onlyde-partitioned (block 52) and then fed to the message I/O (block 53).

The implementation of the retransmission protocol as described in theabove embodiments, may be supplemented with other techniques, which areknown per se in the art. E.g., decision feedback equalization (DFE) anda fractionally spaced equalizer (FSE) may be used to equalize amulti-path faded signal and to provide additional narrowbandinterference cancellation. Also, antenna diversity schemes and dynamicfrequency (re-)allocation protocols may be added to the presentcommunication system. Also, error concealment may be applied to furtherimprove the robustness and efficiency of the communication system. Intotal, a very robust communication system against interference andfading may be achieved, without requiring channel coding/decodingtechniques, interleaving/de-interleaving techniques, or frequencyhopping techniques. Known systems use forward error correction (FEC)schemes, with or without interleaving. Together with block interleavingagainst burst errors, (from interference and fading), a significant gaincan be achieved in bit error rate and frame error rate. However, thiscomes with a number of disadvantages, such as required bufferingcapacity, latency increase, and processing and memory requirements.

In the following, some of the blocks depicted in FIGS. 5 and 6 will beexplained in more detail. It is noted that these implementations ofspecific parts of the transmission system may be applied as such, i.e.in other applications than the automatic retransmission protocolimplementation described above.

In order to increase the capacity of the system, compression anddecompression of data packets (blocks 24,47) may be used. In aparticular embodiment of the present invention, the first transmissionof an audio data packet is not compressed. Upon failed (or repetitivelyfailed) reception of this packet the destination requestsretransmission. The source may then retransmit this packet withcompression.

The error concealment technique (block 48 in FIG. 6) may be implementedusing the following particulars. A (repeatedly) failed retransmissionwill eventually result in audible deformation of the received audiosignal. Using the error concealment technique as described, applying awindowing function, it is possible to fade in and fade out the audiosignal of a recently received audio block. This way, the resulting audiosignal at the receiver end will not be distorted in an unpleasantmanner.

In the present audio transmission system, an air frame failure mayoccur, and an audio block failure. A failed air frame will beretransmitted using the present invention, until a certain number ofretries is reached, after which the air frame will be marked as lost.The audio blocks that depend on this particular air frame are thenmarked as lost as well. A delayed audio block, which is still present ina digital memory of the receiver unit, will replace the lost audioblock. By applying a smooth transition window between the current anddelayed audio blocks, it is possible to prevent ticks in the eventualaudio signal. When more than one audio block is lost, the received audiosignal will fade out to a zero signal (absolute silence).

In FIG. 7, a hardware implementation module 70 of the error concealmenttechnique is shown. The input data, comprising audio blocks, is dividedin two branches. In the normal situation, the audio block is onlydelayed with the window size (block 73) to allow synchronization withthe window multiplier to be discussed later. The selection unit 79 willpass the (delayed) audio block to the output when no lost audio block isdetected. Whenever a lost audio block is present, the selection unit 79will pass the summation of the two branches of FIG. 8 (adder element78). The upper branch represents the current audio block, which ismultiplied by a negative window function 77 in multiplier 75. The lowerbranch first comprises a delay element 71, which delays the audio blockwith exactly an audio block period, and a window delay element 72. Afterthat the (previous) audio block is multiplied by a window function 76and passed to the adder element 78. The window functions 76, 77 may e.g.be cosine functions.

The transmitter arrangement 20 (FIG. 5) may comprise an optionalscrambling unit 31, and the receiver arrangement 40 (FIG. 6) maycomprise a corresponding de-scrambling unit 43. The scrambling unit 31pseudo-randomly may change the data differently each (re-)transmission,and the de-scrambling unit 43 de-scrambles the data signalcorrespondingly. The de-scrambling unit 43 may further comprise asoft-valued pre-detection accumulator. The rationale behind this is thatretransmitted data packets consist of the same data. If theseretransmissions occur within the coherence time of the radio channel,the bit error distribution in the packet will not vary significantlybetween retransmissions. By scrambling each retransmission in adifferent way, the bit error distribution is randomized, allowing forsensitivity gain by means of pre-detection integration. The detectionprocedure is then as follows: A received packet is checked for errors,e.g. with a CRC check. If the packet is in error, the descrambled softdata originating from the demodulator, is added, in the pre-detectionintegrator, to previously received packets with the same packet ID. Thispacket ID is identical for (re)transmissions of the same packet. Thedata in the soft value integrator is detected, and the resulting packetis checked for errors. If the packet is error-free, the transmission isconsidered successful, and no retransmission is requested. Also, thescrambling will result in a flat power density, which may be importantfor compliance with regulatory standards.

The present communication system may comprise multiple central units 10,which then are usually co-located. Mobile units 12 will be scatteredthroughout the coverage area, resulting in near-far problems withrespect to signal power, but mainly in the up-link part of the signalonly. In known power control methods, a fast inner loop is used and aslowly reacting outer loop. The inner loop takes into account a signalcharacteristic, such as signal-to-noise ratio (SNR), received signalstrength indicator (RSSI) or bit-error-rate (BER), in a single slot, andcompares this characteristic to a target level (or threshold value).Depending on the result of the comparison, the transmission power isamended, usually using small up or down steps. The target level is setby the outer loop, e.g. based on average checksum or BER measurements.

In the present communication system, the transmitter immediatelyincreases the transmitting power when receiving a (negative) acknowledgepacket indicating a data packet has been received with error. When apacket is received correctly, the RSSI is compared to a target level,and depending on the comparison, the transmitter power level is eitherstepped up or down. This target level may e.g. be determined in an outerloop, as mentioned above, or may be a default value.

Although the above embodiment has been explained using RF communicationsin the ISM band, it is of course possible to implement the invention inother RF frequency bands, or even using other wireless techniques, suchas infrared. The examples of the invention as described above relate toa time division multiplexing (TDM) scheme, in which a data frame isdivided in an number of time slots. However, the present application mayalso be implemented in multiplexing schemes of other types, such as, butnot limited to frequency division multiplexing, code divisionmultiplexing, etc.

The invention claimed is:
 1. A method for transmitting at least one datastream from a source to at least one destination over a communicationchannel, wherein the at least one data stream comprises a sequence of aplurality of data packets, and wherein the method comprises:transmitting the at least one data stream over the communication channelusing a signal comprising data frames according to a protocol, whereineach data frame comprises N fixed slots with a single negativeacknowledge sub-slot following each of the N fixed slots, and at leastone freely allocatable slot, wherein each freely allocatable slotcomprises N negative acknowledge sub-slots following the freelyallocatable slot for indicating in which of the N fixed slots a specificdata packet was not received; and retransmitting a specific data packet,which is not properly received by the at least one destination, from thesource using one of the at least one freely allocatable slot, whereinthe specific data packet is transmitted uncompressed in the step oftransmitting, and compressed in the step of retransmitting.
 2. Themethod according to claim 1, where the step of retransmitting comprisesretransmitting the specific data packet for a specific data stream inthe fixed slot of that specific data stream in a next data frame.
 3. Themethod according to claim 1, wherein each data frame comprises adown-link part and an up-link part, each having at least one fixed slotand at least one freely allocatable slot.
 4. The method according toclaim 1, wherein a data packet has a predefined duration, the predefinedduration being small compared to non-transmitting gaps of possibleinterfering sources.
 5. The method according to claim 4, furthercomprising detecting the non-transmitting gaps by carrier sense/detecttechniques, and synchronizing the data stream transmission to thedetected non-transmitting gaps.
 6. The method according to claim 1,further comprising receiving acknowledgement of a received data packetfrom the at least one destination, and retransmitting a not properlyreceived data packet within the same data frame.
 7. The method accordingto claim 6, wherein the acknowledgment comprises a pseudo-noise code. 8.The method according to claim 1, further comprising allocating at leastone freely allocatable slot to transmit a control data message.
 9. Themethod according to claim 8, further comprising using a back-offmechanism for transmitting the control data message for multiple access.10. The method according to claim 1, further comprising applying antennaand/or frequency diversity.
 11. The method according to claim 1, furthercomprising: converting received data packets into an audio signal;replacing a missing data packet with an earlier received data packet;and smoothing a transition between the earlier received data packet andthe replaced data packet.
 12. The method according to claim 11, whereinthe step of smoothing comprises using a raised cosine filter function tosmooth the transition between the earlier received data packet and thereplaced data packet.
 13. The method according to claim 1, furthercomprising scrambling a retransmitted data packet before retransmissionusing a pseudo-randomly varying scrambling technique, and descramblingthe retransmitted data packet upon reception.
 14. The method accordingto claim 13, further comprising integrating multiple retransmitted datapackets.
 15. The method according to claim 1, further comprising thesource increasing its transmission power upon determination that a datapacket is not properly received by the destination.
 16. The methodaccording to claim 1, further comprising the source comparing a receivedsignal strength to a threshold value, and either: (a) decreasing itstransmission power by a predefined step if the threshold value isexceeded, or (b) increasing its transmission power by a predefined stepotherwise.
 17. The method according to claim 16, wherein the thresholdvalue is adaptively controlled by an outer control loop.
 18. The methodaccording to claim 1, wherein a data packet comprises one or more of apreamble, a header, at least one packet of control message data, and atleast one packet of application data from at least one input.
 19. Atransmitter arrangement for use in a system for transmitting at leastone data stream from a source to at least one destination over acommunication channel, the transmitter arrangement comprising: inputprocessing means for assembling data packets for each of the at leastone data stream a multiplexer arranged to receive data packets from theinput processing means for each of the at least one data stream, whereinthe multiplexer is configured to assemble a plurality of data frames tobe transmitted over the communication channel, wherein each data framecomprises N fixed slots with a single negative acknowledge sub-slotfollowing each of the N fixed slots, and at least one freely allocatableslot wherein each freely allocatable slot comprises N negativeacknowledge sub-slots following the freely allocatable slot forindicating in which of the N fixed slots a specific data packet was notreceived; retransmission control means connected to the multiplexer andconfigured to retransmit a specific data packet, which is not properlyreceived by the at least one destination, using one of the at least onefreely allocatable slot; and a compressor connected to theretransmission control means to enable the specific data packet to betransmitted uncompressed in a first transmission, and compressed in atleast one of the retransmissions.
 20. The transmitter arrangementaccording to claim 19, wherein at least one of the multiplexer and theretransmission control means are further configured to execute a methodin which the specific data packet for a specific data stream isretransmitted in the fixed slot of that specific data stream in a nextdata frame.
 21. The transmitter arrangement according to claim 19,further comprising a scrambler for scrambling a retransmitted datapacket before retransmission using a pseudo-randomly varying scramblingtechnique.
 22. The transmitter arrangement according to claim 19,further comprising a modulator.
 23. A receiver arrangement for receivingat least one data stream transmitted by the transmitter arrangementaccording to claim
 19. 24. The receiver arrangement according to claim23, comprising audio processing electronics.
 25. The receiverarrangement according to claim 23, further comprising a descrambler fordescrambling a retransmitted data packet upon reception.
 26. Thereceiver arrangement according to claim 25, further comprising apre-detection accumulator for integrating multiple retransmitted datapackets.